Various spelling fixes

Signed-off-by: wm4 <wm4@nowhere>

8 files changed, 13 insertions, 13 deletions

diff --git a/audio/chmap.h b/audio/chmap.h
index adb7481665..ba1072547b 100644
--- a/audio/chmap.h
+++ b/audio/chmap.h
@@ -47,7 +47,7 @@ enum mp_speaker_id {
     MP_SPEAKER_ID_TBL,          // TOP_BACK_LEFT
     MP_SPEAKER_ID_TBC,          // TOP_BACK_CENTER
     MP_SPEAKER_ID_TBR,          // TOP_BACK_RIGHT
-     // Inofficial/libav* extensions
+     // Unofficial/libav* extensions
     MP_SPEAKER_ID_DL = 29,      // STEREO_LEFT (stereo downmix special speakers)
     MP_SPEAKER_ID_DR,           // STEREO_RIGHT
     MP_SPEAKER_ID_WL,           // WIDE_LEFT
diff --git a/audio/filter/af.c b/audio/filter/af.c
index 5a686e813c..e67fc29203 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -646,7 +646,7 @@ struct af_instance *af_add(struct af_stream *s, char *name, char *label,
         return NULL;
     new->label = talloc_strdup(new, label);
 
-    // Reinitalize the filter list
+    // Reinitialize the filter list
     if (af_reinit(s) != AF_OK) {
         af_remove_by_label(s, label);
         return NULL;
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
index 4344766349..472758c4c7 100644
--- a/audio/filter/af_drc.c
+++ b/audio/filter/af_drc.c
@@ -131,7 +131,7 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c)
     data[i] = tmp;
   }
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -168,7 +168,7 @@ static void method1_float(af_drc_t *s, struct mp_audio *c)
   for (i = 0; i < len; i++)
     data[i] *= s->mul;
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -216,7 +216,7 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
     data[i] = tmp;
   }
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -262,7 +262,7 @@ static void method2_float(af_drc_t *s, struct mp_audio *c)
   for (i = 0; i < len; i++)
     data[i] *= s->mul;
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
index f2613530e5..6020d9d98e 100644
--- a/audio/filter/af_export.c
+++ b/audio/filter/af_export.c
@@ -167,7 +167,7 @@ static int filter(struct af_instance *af, struct mp_audio *data)
     return 0;
   struct mp_audio*      c   = data;          // Current working data
   af_export_t*  s   = af->priv;     // Setup for this instance
-  int16_t*      a   = c->planes[0];          // Incomming sound
+  int16_t*      a   = c->planes[0];          // Incoming sound
   int           nch = c->nch;        // Number of channels
   int           len = c->samples*c->nch; // Number of sample in data chunk
   int           sz  = s->sz;         // buffer size (in samples)
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
index 94a1599cd0..3c8a89c665 100644
--- a/audio/filter/af_hrtf.c
+++ b/audio/filter/af_hrtf.c
@@ -206,7 +206,7 @@ static inline void matrix_decode(short *in, const int k, const int il,
       information about Lt, Rt correlation.  This effectively reshapes
       the front and rear "cones" to concentrate Lt + Rt to C and
       introduce Lt - Rt in L, R. */
-   /* 0.67677 is the emprical lower bound for lpr_gain. */
+   /* 0.67677 is the empirical lower bound for lpr_gain. */
    c_gain = 8 * (*adapt_lpr_gain - 0.67677);
    c_gain = c_gain > 0 ? c_gain : 0;
    /* c_gain should not be too high, not even reaching full
diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c
index bd54dbb267..edde6a68b1 100644
--- a/audio/filter/af_ladspa.c
+++ b/audio/filter/af_ladspa.c
@@ -144,7 +144,7 @@ static int af_ladspa_parse_plugin(struct af_instance *af) {
     LADSPA_PortRangeHint hint;
 
     if (!setup->libhandle)
-        return AF_ERROR; /* only call parse after a succesful load */
+        return AF_ERROR; /* only call parse after a successful load */
     if (!setup->plugin_descriptor)
         return AF_ERROR; /* same as above */
 
diff --git a/audio/mixer.c b/audio/mixer.c
index 7ecd97449d..29727918f6 100644
--- a/audio/mixer.c
+++ b/audio/mixer.c
@@ -227,7 +227,7 @@ void mixer_setbalance(struct mixer *mixer, float val)
         return;
     }
 
-    /* make all other channels pass thru since by default pan blocks all */
+    /* make all other channels pass through since by default pan blocks all */
     for (int i = 2; i < AF_NCH; i++) {
         float level[AF_NCH] = {0};
         level[i] = 1.f;
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index 9d216e673b..8b1e10a10b 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -288,7 +288,7 @@ static int InitDirectSound(struct ao *ao)
     /* Set DirectSound Cooperative level, ie what control we want over Windows
      * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
      * settings of the primary buffer, but also that only the sound of our
-     * application will be hearable when it will have the focus.
+     * application will be audible when it will have the focus.
      * !!! (this is not really working as intended yet because to set the
      * cooperative level you need the window handle of your application, and
      * I don't know of any easy way to get it. Especially since we might play
@@ -616,9 +616,9 @@ static int check_free_buffer_size(struct ao *ao)
     space = p->buffer_size - (p->write_offset - play_offset);
     // |              | <-- const --> |                |                 |
     // buffer start   play_cursor     write_cursor     p->write_offset   buffer end
-    // play_cursor is the actual postion of the play cursor
+    // play_cursor is the actual position of the play cursor
     // write_cursor is the position after which it is assumed to be save to write data
-    // p->write_offset is the postion where we actually write the data to
+    // p->write_offset is the position where we actually write the data to
     if (space > p->buffer_size)
         space -= p->buffer_size;                        // p->write_offset < play_offset
     // Check for buffer underruns. An underrun happens if DirectSound